yes @arsperger, you are correct. here is the wrong Route header used in ACK message.
Wrong route header used because of Kamailio inserter wrong Record-Route header.

WebRTC client really connected to the socket tls:3.236.25.254:7001 but Record-Route header generated like

INVITE sips:safarov@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss SIP/2.0
Record-Route: <sips:3.236.25.254;transport=ws;r2=on;lr=on;ftag=j4aU17p3BvH0e;did=698.6e42>
Record-Route: <sips:[2600:1f18:578:5701::7e];transport=tcp;r2=on;lr=on;ftag=j4aU17p3BvH0e;did=698.6e42>
Record-Route: <sip:[2600:1f14:6d8:5408::100]:5080;transport=tcp;r2=on;lr=on;ftag=j4aU17p3BvH0e>
Record-Route: <sip:[2600:1f14:6d8:5408::100];transport=tcp;r2=on;lr=on;ftag=j4aU17p3BvH0e>
Via: SIP/2.0/WSS 3.236.25.254;branch=z9hG4bKa53e.f43ae42f2fefe414dabc1a6739b19bbe.0;i=3

Here is top Record-Route should contain port "7001" but really is not.

I was replaced use of record_route by

record_route_preset("3.236.25.254:7001;transport=ws;r2=on", "[2600:1f18:578:5701::7e];transport=tcp;r2=on");

This alow me to specify which sockets need to use for call dialog and now ACK properly delivered to WebRTC client.

The issue about missed port in Record-Route header will be created.
This ticket is resolved.

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